Method and apparatus for transmitting coded audio signals through a transmission channel with limited bandwidth

ABSTRACT

A digital audio transmitter system capable of transmitting high quality, wideband speech over a transmission channel with a limited bandwidth such as a traditional telephone line. The digital audio transmitter system includes a coder for coding an input audio signal to a digital signal having a transmission rate that does not exceed the maximum allowable transmission rate for traditional telephone lines and a decoder for decoding the digital signal to provide an output audio signal with an audio bandwidth of wideband speech. A coder and a decoder may be provided in a single device to allow two-way communication between multiple devices.

CROSS REFERENCE TO RELATED APPLICATIONS

This is a continuation of Ser. No. 09/595,521, filed Jun. 16, 2000,issued as U.S. Pat. No 6,373,927 which is a continuation of Ser. No.08/988,709, filed Dec. 11, 1997, issued as U.S. Pat. No. 6,128,374,which is a continuation of Ser. No. 08/419,199, filed Apr. 10, 1995,issued as U.S. Pat. No. 5,706,335.

FIELD OF THE INVENTION

The present invention relates generally to an apparatus and method fortransmitting audio signals and pertains, more specifically, to anapparatus and method for transmitting a high quality audio signal, suchas wideband speech, through a transmission channel having a limitedbandwidth or transmission rate.

BACKGROUND OF THE INVENTION

Human speech lies in the frequency range of approximately 7 Hz to 10kHz. Because traditional telephone systems only provide for thetransmission of analog audio signals in the range of about 300 Hz to3400 Hz or a bandwidth of about 3 kHz (narrowband speech), certaincharacteristics of a speaker's voice are lost and the voice soundssomewhat muffled. A telephone system capable of transmitting an audiosignal approaching the quality of face-to-face speech requires abandwidth of about 6 kHz (wideband speech).

Known digital transmission systems are capable of transmitting widebandspeech audio signals. However, in order to produce an output audiosignal of acceptable quality with a bandwidth of 6 kHz, these digitalsystems require a transmission channel with a transmission rate thatexceeds the capacity of traditional telephone lines. A digital systemtransmits audio signals by coding an input audio signal into a digitalsignal made up of a sequence of binary numbers or bits, transmitting thedigital signal through a transmission channel, and decoding the digitalsignal to produce an output audio signal. During the coding process thedigital signal is reduced or compressed to minimize the necessarytransmission rate of the signal. One known method for compressingwideband speech is disclosed in Recommendation G.722 (CCITT, 1988). Asystem using the compression method described in G.722 still requires atransmission rate of at least 48 kbit/s to produce wideband speech of anacceptable quality.

Because the maximum transmission rate over traditional telephone linesis 28.8 kbit/s using the most advanced modem technology, alternativetransmission channels such as satellite or fiber optics would have to beused with an audio transmission system employing the data compressionmethod disclosed in G.722. Use of these alternative transmissionchannels is both expensive and inconvenient due to their limitedavailability. While fiber optic lines are available, traditional coppertelephone lines now account for an overwhelming majority of existinglines and it is unlikely that this balance will change anytime in thenear future. A digital phone system capable of transmitting widebandspeech over existing transmission rate limited telephone phone lines istherefore highly desirable.

OBJECTS OF THE INVENTION

The disclosed invention has various embodiments that achieve one or moreof the following features or objects:

An object of the present invention is to provide for the transmission ofhigh quality wideband speech over existing telephone networks,

A further object of the present invention is to provide for thetransmission of high quality audio signals in the range of 20 Hz to atleast 5,500 Hz over existing telephone networks.

A still further object of the present invention is to accomplish datacompression on wideband speech signals to produce a transmission rate of28.8 kbit/s or less without significant loss of audio quality.

A still further object of the present invention is to provide a devicewhich allows a user to transmit and receive high quality wideband speechand audio over existing telephone networks.

A still further object of the present invention is to provide a portabledevice which is convenient to use and allows ease of connection toexisting telephone networks.

A still further object of the present invention is to provide a devicewhich is economical to manufacture.

A still further object of the present invention is to provide easy andflexible programmability.

SUMMARY OF THE INVENTION

In accordance with the present invention, the disadvantages of the priorart have been overcome by providing a digital audio transmitter systemcapable of transmitting high quality, wideband speech over atransmission channel with a limited bandwidth such as a traditionaltelephone line.

More particularly, the digital audio transmitter system of the presentinvention includes a coder for coding an input audio signal to a digitalsignal having a transmission rate that does not exceed the maximumallowable transmission rate for traditional telephone lines and adecoder for decoding the digital signal to provide an output audiosignal with an audio bandwidth of wideband speech. A coder and a decodermay be provided in a single device to allow two-way communicationbetween multiple devices. A device containing a coder and a decoder iscommonly referred to as a CODEC (COder/DECoder).

These and other objects, advantages and novel features of the presentinvention, as well as details of an illustrative embodiment thereof,will be more fully understood from the following description and fromthe drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of a digital audio transmission systemincluding a first CODEC and second CODEC in accordance with the presentinvention.

FIG. 2 is a block diagram of a CODEC of FIG. 1.

FIG. 3 is a block diagram of an audio input/output circuit of a CODEC.

FIG. 4 is a detailed circuit diagram of the audio input portion of FIG.3.

FIG. 5 is a detailed circuit diagram of the level LED's portion of FIG.3.

FIG. 6 is a detailed circuit diagram of the headphone amp portion ofFIG. 3.

FIG. 7 is a block diagram of a control processor of a CODEC.

FIG. 8 is a detailed circuit diagram of the microprocessor portion ofthe control processor of FIG. 7.

FIG. 9 is a detailed circuit diagram of the memory portion of thecontrol processor of FIG. 7.

FIG. 10 is a detailed circuit diagram of the dual UART portion of thecontrol,processor of FIG. 7.

FIG. 11 is a detailed circuit diagram of the keypad, LCD display andinterface portions of the control processor of FIG. 7.

FIG. 12 is a block diagram of an encoder of a CODEC.

FIG. 13 is a detailed circuit diagram of the encoder digital signalprocessor and memory,portions of the encoder of FIG. 12.

FIG. 14 is a detailed circuit diagram of the clock generator portion ofthe encoder of FIG. 12.

FIG. 15 is a detailed circuit diagram of the Reed-Soloman encoder anddecoder portions of FIGS. 12 and 16.

FIG. 16 is a block diagram of a decoder of a CODEC.

FIG. 17 is a detailed circuit diagram of the encoder digital signalprocessor and memory portions of the decoder of FIG. 16.

FIG. 18 is a detailed circuit diagram of the clock generator portion ofthe decoder of FIG. 16.

FIG. 19 is a detailed circuit diagram of the analog/digital converterportion of the encoder of FIG. 12.

FIG. 20 is a detailed circuit diagram of the digital/analog converterportion of the decoder of FIG. 16.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT

A digital audio transmission system 10, as shown in FIG. 1, includes afirst CODEC (COder/DECoder) 12 for transmitting and receiving a widebandaudio signal such as wideband speech to and from a second CODEC 14 via atraditional copper telephone line 16 and telephone network 17. Whentransmitting an audio signal, the first CODEC 12 performs a codingprocess on the input analog audio signal which includes converting theinput audio signal to a digital signal and compressing the digitalsignal to a transmission rate of 28.8 kbit/s or less. The preferredembodiment compresses the digital using a modified version of theISO/MPEG (International Standards Organization/Motion Picture ExpertGroups) compression scheme according to the software routine disclosedin the microfiche software appendix filed herewith. The coded digitalsignal is sent using standard modem technology via the telephone line 16and telephone network 17 to the second CODEC 14, The second CODEC 14performs a decoding process on the coded digital signal by correctingtransmission errors, decompressing the digital signal and reconvertingit to produce an output analog audio signal.

FIG. 2 shows a CODEC 12 which includes an analog mixer 20 for receiving,amplifying, and mixing an input audio signal through a number of inputlines. The input lines may include a MIC line 22 for receiving an analogaudio signal from a microphone and a generic LINE 24 input for receivingan analog audio signal from an audio playback device such as a tapedeck. The voltage level of an input audio signal on either the MIC line22 or the generic LINE 24 can be adjusted by a user of the CODEC 12 byadjusting the volume controls 26 and 28 When the analog mixer 20 isreceiving an input signal through both the MIC line 22 and the genericLINE 24, the two signals will be mixed or combined to produce a singleanalog signal. Audio level LED's 30 respond to the voltage level of amixed audio signal to indicate when the voltage exceeds a desiredthreshold level. A more detailed description of the analog mixer 20 andaudio level LED's 30 appears below with respect to FIGS. 3 and 4.

The combined analog signal from the analog mixer 20 is sent to theencoder 32 where the analog signal is first converted to a digitalsignal. The sampling rate used for the analog to digital conversion ispreferably one-half the transmission rate of the signal which willultimately be transmitted to the second CODEC 14 (shown in FIG. 1).After analog to digital conversion, the digital signal is thencompressed using a modified version of the ISO/MPEG algorithm. TheISO/MPEG compression algorithm is modified to produce a transmissionrate of 28.8 kbit/s. This is accomplished by the software routine thatis disclosed in the software appendix.

The compressed digital signal from the encoder 32 is then sent to anerror protection processor 34 where additional error protection data isadded to the digital signal. A Reed-Solomon error protection format isused by the error protection processor 34 to provide both burst andrandom error protection. The error protection processor 34 is describedbelow in greater detail with respect to FIGS. 12 and 15.

The compressed and error protected digital signal is then sent to ananalog modem 36 where the digital signal is converted back to an analogsignal for transmitting. As shown in FIG. 1, this analog signal is sentvia a standard copper telephone line 16 through a telephone network 17to the second CODEC 14. The analog modem 36 is preferably a V.34synchronous modem. This type of modem is commercially available.

The analog modem 36 is also adapted to receive an incoming analog signalfrom the second CODEC 14 (or another CODEC) and reconvert the analogsignal to a digital signal. This digital signal is then sent to an errorcorrection processor 38 where error correction according to aReed-Soloman format is performed.

The corrected digital signal is then sent to a decoder 40 where it isdecompressed using the modified version of the ISO/MPEG algorithm asdisclosed in the software appendix. After decompression the digitalsignal is converted to an analog audio signal. A more detaileddescription of the decoder 40 appears below with respect to FIGS. 7, 16,17 and 18. The analog audio signal may then be perceived by a user ofthe CODEC 12 by routing the analog audio signal through a headphone amp42 wherein the signal is amplified. The volume of the audio signal atthe headphone output line 44 is controlled by volume control 46.

The CODEC 12 includes a control processor 48 for controlling the variousfunctions of the CODEC 12 according to software routines stored inmemory 50. A more detailed description of the structure of the controlprocessor appears below with respect to FIGS. 7, 8, 9, 10, and 11. Onesoftware routine executed by the control processor allows the user ofthe CODEC 12 to initiate calls and enter data such as phone numbers.When a call is initiated the control processor sends a signal includingthe phone number to be dialed to the analog modem 36. Data entry isaccomplished via a keypad 52 and the entered data may be monitored byobservation of an LCD 54. The keypad 52 also includes keys for selectingvarious modes of operation of the CODEC 12. For example, a user mayselect a test mode wherein the control processor 48 controls the signalpath of the output of the encoder to input of decoder to bypass thetelephone network allows testing of compression and decompressionalgorithms and their related hardware Also stored in memory 50 is thecompression algorithm executed by the encoder 32 and the decompressionalgorithm executed by the decoder 40.

Additional LED's 56 are controlled by the control processor 48 and mayindicate to the user information such as “bit synchronization” (achievedby the decoder) or “power on”. An external battery pack 58 is connectedto the CODEC 12 for supplying power.

FIG. 3 shows a lower level block diagram of the analog mixer 20, audiolevel LED's 30 and analog headphone amp, 42 as shown in FIG. 2. FIGS. 4,5 and 6 are the detailed circuit diagrams corresponding to FIG. 3.

Referring to FIGS. 3 and 4, line input 210 is an incoming line levelinput signal while mic input 220 is the microphone level input. Thesesignals are amplified by a line amp 300 and a mic amp respectively andtheir levels are adjusted by line level control 304 and mic levelcontrol 306 respectively. The microphone and line level inputs are fedto the input mixer 308 where they are mixed and the resulting combinedaudio input signal 310 is developed.

Referring now to FIGS. 3 and 5, the audio input signal 310 is to thenormal and overload signal detectors, 312 and 314 respectively, wheretheir level is compared to a normal threshold which defines a normalvolume level and a clip threshold 318 which defines an overload volumelevel. When the audio input signal 310 is at a normal volume level aNORM LED 320 is lighted. when the audio input signal 310 is at anoverload volume level a clip LED 322 is lighted.

Referring now to FIGS. 3 and 6, the audio input signal 310 is fed intothe record monitor level control 324, where its level is adjusted beforebeing mixed with the audio output signal 336 from the digital/analogconverter 442 (shown in FIGS. 16 and 20). The audio output signal 336 isfed to the local monitor level control 326 before it is fed into theheadphone mixer amplifier 334. The resulting output signal from theheadphone mixer amplifier 334 goes to a headphone output connector 338on the exterior of the CODEC 12 where a pair of headphones may beconnected.

The audio input signal 310 and audio output signal 336 are fed to recordmix control 328 which is operable by the user. The output of thiscontrol is fed to a mix level control 330 (also operable by a user) andthen to the record output amplifier 332. The resulting output signal ofthe record output amplifier 332 goes to a record output 340 on theexterior of the CODEC 12.

FIG. 7 shows a lower level block diagram of the control processor 48(shown in FIG. 2). The encoder 406 (referenced as number 32 in FIG. 2)is further described in FIG. 12 while the decoder 416 (referenced asnumber 40 in FIG. 2) is refined in FIG. 16. FIGS. 8, 9, 10, 11, 13, 14,15, 17, 18, 19 and 20 are detailed circuit diagrams.

Referring to FIGS. 7 and 8 the microprocessor 400 is responsible for thecommunication between the user, via keypad 412 and LCD display 414, andthe CODEC 12. The keypad 412 is used to input commands to the systemwhile the LCD display 414, is used to display the responses of thekeypad 412 commands as well as alert messages generated by the CODEC 12.

Referring now to FIGS. 7 and 9, the RAM (random access memory) 402 isused to hold a portion of the control processor control softwareroutines. The flash ROM (read only memory) 404 holds the softwareroutine (disclosed in the software appendix) which controls the modifiedISO/MPEG compression scheme performed by encoder DSP 406 and the.modified ISO/MPEG decompression scheme performed by the decoder DSP 416,as well as the remainder of the control processor control softwareroutines.

Referring now to FIGS. 7 and 10, the dual UART (universal asynchronousreceiver/transmitter) 408 is used to provide asynchronous input/outputfor the control processor 48. The rear panel remote control port 409 andthe rear panel RS232 port 411 are used to allow control by an externalcomputer. This external control can be used in conjunction with orinstead of the keypad 412 and/or LCD display 414.

Referring now to FIGS. 7 and 11, the programmable interval timer circuit410 is used to interface the control processor with the keypad and LCDdisplay.

Referring now to FIGS. 7, 8 and 13, the encoder DSP (digital signalprocessor) 434 receives a digital pulse code modulated signal 430 fromthe analog/digital converter 450. The encoder DSP 434 performs themodified ISO/MPEG compression scheme according to the software routine(described in the software appendix) stored in RAM memory 436 to producea digital output 418.

The A/D clock generation unit 439 is shown in FIGS. 12 and 14. Thefunction of this circuitry is to provide all the necessary timingsignals for the analog digital converter 450 and the encoder DSP 434.

The Reed-Soloman error correction encoding circuitry 438 is shown inFIGS. 12 and 15. The function of this unit is to add parity informationto be used by the Reed-Soloman decoder 446 (also shown in FIG. 16) torepair any corrupted bits received by the Reed-Soloman decoder 446. TheReed-Soloman corrector 438 utilizes a shortened Reed-Soloman GF(256)code which might contain, for example, code blocks containing 170eight-bit data words and 8 eight-bit parity words.

Referring now to FIGS. 7, 16 and 17, the decoder DSP 440 receives adigital input signal 422 from the modem 36 (shown in FIG. 2) The decoderDSP 440 performs the modified ISO/MPEG decompression scheme according tothe software routine (described in the software appendix) stored in RAMmemory 444 to produce a digital output to be sent to the digital/analogconverter 442.

The D/A clock generation unit 448 is shown in FIGS. 16 and 18. Thefunction of this circuitry is to provide all the necessary timingsignals for the digital/analog converter 442 and the decoder DSP 440.

The analog/digital converter 450, shown in FIGS. 12 and 19, is used toconvert the analog input signal 310 into a PCM digital signal 430.

The digital/analog converter 442, shown in FIGS. 16 and 20 is used toconvert the PCM digital signal from the decoder DSP 440 into an analogaudio output signal 336.

The Reed-Soloman error correction decoding circuitry 446, shown in FIGS.15 and 16, decodes a Reed-Soloman coded signal to correct errorsproduced during transmission of the signal through the modem 36 (shownin FIG. 2) and telephone network.

Another function contemplated by this invention is to allow real time,user operated adjustment of a number of psycho-acoustic parameters ofthe ISO/MPEG compression/decompression scheme used by the CODEC 12. Amanner of implementing this function is described in applicant'sapplication entitled “System For Adjusting Psycho-Acoustic Parameters InA Digital Audio Codec” which is being filed concurrently herewith (suchapplication and related Software Appendix are hereby incorporated byreference). Also, applicants application entitled “System ForCompression And Decompression Of Audio Signals For Digital Transmission”and related Software Appendix which are being filed concurrentlyherewith are hereby incorporated by reference. This

this invention has been described above with reference to a preferredembodiment. Modifications and variations may become apparent to oneskilled in the art upon reading and understanding this specification. Itis intended to include all such modifications and alterations within thescope of the appended claims.

What is claimed is:
 1. An portable audio transmission system comprising:a coder for coding an entire input audio signal into a digital signal ina single encoding process at a transmission speed including 28.8 kbit/sto be transmitted through a traditional analog copper telephone linegenerally supporting a digital signal transmission rate of at least 28.8kbit/s; and a decoder for decoding the digital signal that is receivedfrom a telephone network to provide an output audio signal with afrequency range of greater than 4 kilohertz.
 2. A portable CODEC fortransmitting high quality audio signals over a standard telephone linehaving a limited bandwidth and maximum transmission rate, said portableCODEC comprising: a single portable housing; a memory within the housingand storing a lossy audio compression routine; an encoder within thehousing and including a program to convert an entire audio input signalto a digital input signal in a single encoding process at a samplingrate and encode said digital input signal based on said lossycompression routine stored in memory to produce an encoded digitalsignal having a compression ratio with respect to said audio inputsignal; an analog modem within the housing and establishing a connectionwith, and a transmission rate for, a standard telephone line of atelephone network, said modem converting said encoded digital signal toan encoded analog output signal and outputting said encoded analogoutput signal at said transmission rate established by said analog modemalong the standard telephone line through the telephone network; and aprocessor within the housing and enabling said analog modem to outputsaid encoded analog output signal at a transmission rate that does notexceed a predetermined transmission rate of the standard telephone line,said standard telephone line generally supporting a transmission rate ofat least 28.8 kbit/s.
 3. A portable CODEC according to claim 2, furthercomprising a clock generator providing synchronous clock signals to saidencoder and analog modem.
 4. A portable CODEC according to claim 2,wherein said processor defines said sampling rate to equal approximatelyone-half of said transmission rate established by said analog modem. 5.A portable CODEC according to claim 2, further comprising a microphoneinput line within the housing whereby said microphone input line mayreceive live, real time analog audio signals.
 6. A portable CODECaccording to claim 2, further comprising an input line within thehousing whereby an analog audio signal may be received from an audioplayback device.
 7. A portable CODEC according to claim 2, furthercomprising a voltage level adjuster in the housing whereby the voltagelevel of an input audio signal on said at least one input line can beadjusted.
 8. A portable CODEC according to claim 2 further comprising ananalog mixer within the housing, said analog mixer receiving, amplifyingand mixing at least two input audio signals to produce said audio inputsignal to said encoder.
 9. A portable CODEC according to claim 2,further comprising a least one audio level display indicator within thehousing and indicating when a voltage level of said single input signalexceeds a threshold level.
 10. A protable CODEC according to claim 8,wherein said analog mixer comprises: line amplifiers amplifying inputaudio signals on at least two input lines; line level controllers,connected to said amplifiers, adjustable by a user, said levelcontrollers controlling an output voltage to which input audio signalsare amplified by said amplifier, and an input mixer mixing amplifiedaudio signals output by said level controllers to produce said audioinput signal.
 11. A portable CODEC according to claim 8, wherein saidanalog mixer comprises: normal and overload signal detectors comparingsaid single combined audio input signal with normal and clip thresholdsdefining normal and overload volume levels, respectively, and normal andoverload displays connected to said normal and overload signaldetectors, respectively, said normal display when said audio inputsignal is at said normal threshold, said overload display lighting whensaid single combined audio input signal is at said overload threshold.12. A portable CODEC according to claim 2 wherein said encoder encodessaid digital input signal based on parameters stored in memory thatproduce encoded digital signals having a bandwidth range ofapproximately 20 Hz to 5,500 Hz.
 13. A portable CODEC according to claim2 wherein said encoder encodes said digital input signal based onparameters stored in memory that produce encoded digital signals havinga bandwidth range of approximately 300 Hz to 3,000 Hz.
 14. A portableCODEC according to claim 2 wherein said encoder encodes said digitalinput signal based on an ISO/MPEG Layer II compression routine havingpredefined psycho-acoustic parameter levels that produce an encodeddigital signal having a bandwidth range of approximately 20 Hz to 5,500Hz.
 15. A portable CODEC according to claim 2 further comprising: anerror protection processor adding error protection date to said encodeddigital signal based on a predefined error protection format to producean encoded and error protected digital signal, said analog modemoutputting said encoded and error protected digital signal at saidoutput signal.
 16. A portable CODEC according to claim 15 wherein saidpredefined error protection format is a Reed-Solomon error protectionformat, said error protection processor providing both burst and randomerror protection.
 17. A portable CODEC according to claim 2 wherein saidanalog modem receives a single incoming encoded analog signal from saidstandard telephone line on said telephone network, said modem convertingsaid single incoming encoded analog signal to an incoming encodeddigital signal.
 18. A portable CODEC according claim 17 wherein saidincoming encoded analog signal contains error protection data, saidCODEC further comprising: an error protection processor performing errorcorrection upon said incoming encoded digital signal based on said errorprotection data to produce an incoming error corrected encoded digitalsignal.
 19. A portable CODEC according to claim 18 wherein said errorcorrection processor comprises: an error correction encoding circuitgenerating parity information based on said incoming encoded digitalsignal; and a Reed-Solomon encoder receiving and preparing corrupteddata bits in said incoming encoded digital signal based on said parityinformation to correct errors produced during transmission through thetelephone network.
 20. A portable CODEC according to claim 19 wherein acode of said Reed-Solomon encoder includes code blocks containingapproximately 178-bit data words and 8-bit parity words.
 21. A portableCODEC according to claim 17 further comprising: a decoder decoding saidincoming encoded digital signal from said analog modem based on a lossydecompression routine stored in memory to provide an analog outputsignal.
 22. A portable CODEC according to claim 21 wherein saidprocessor is selectable by a user between multiple modes of operation,said processor, when in a test mode, bypassing said telephone networkand directing said single encoded digital signal from said encoderdirectly to said decoder to allow testing of said lossy compression andsaid lossy decompression routines in stored memory.
 23. A portable CODECaccording to claims 21 further comprising a clock generator forproviding synchronized clock signals to said encoder and decoder.
 24. Aportable CODEC according to claim 21 wherein said decoder comprises:memory storing an ISO/MPEG decompression routine; and a digital signalprocessor decoding and converting said incoming encoded digital signalbased on said ISO/MPEG decompression routine stored in memory to producesaid analog output signal.
 25. A portable CODEC according to claim 24wherein said decoder further comprises: a D/A converter converting adigital output of said digital signal processor to said analog outputsignal.
 26. A portable CODEC according to claim 25 wherein said decoderfurther comprises a D/A clock generation unit generating synchronoustiming signals for said D/A converter and digital signal processor. 27.A portable CODEC according to claim 2 further comprising: a headphoneamplifier outputting said analog output signal to a headphone outputline; and a volume control controlling the volume of said analog outputsignal at said headphone output line.
 28. A portable CODEC according toclaim 27 wherein said headphone amplifier further comprises: record andlocal monitor level controls receiving and adjusting levels of saidaudio input signal and of said analog output signal from said decoder;and a headphone mixer amplifier mixing output signals of said record andlocal monitored level controls to output a mixed record/local outputsignal at said headphone output line.
 29. A portable CODEC according toclaim 27 further comprising an analog mixer in the housing providing amixed audio signal from multiple analog audio sources and wherein saidheadphone amplifier further comprises: a record mix controller operativeby the user, receiving said mixed audio signal from said analog mixer,said mix controller controlling a level of said audio input signal; anda record output amplifier controlled by said record mix controlleroutputting said audio input signal at a desired level to a recordoutput.
 30. A portable CODEC according to claim 2, wherein saidprocessor comprises: a keypad interface adapted to communicate with akeypad and display respectively; and a microprocessor communicating withthe user through the keypad interface.
 31. A portable CODEC according toclaim 2, further comprising: a keypad entering input commands to saidprocessor; and a display displaying responses to said input commands anddisplaying alert messages.
 32. A portable CODEC according to claim 31,further comprising: a programmable interval timer circuit interfacingsaid processor with said keypad and display.
 33. A portable CODECaccording to claim 32, further comprising: a universal asynchronousreceiver/transmitter providing a synchronous input/output data to saidprocessor from an external computer through a remote control port and aserial port in said receiver/transmitter.
 34. A portable CODEC accordingto claim 2, wherein said encoder comprises: an A/D converter convertingsaid audio input signal to a digital pulse code modulated signal at saidsampling rate; and a digital signal processor encoding said digitalpulse code modulated signal based on a modified ISO/MPEG compressionroutine stored in said memory to produce said encoded signal.
 35. Aportable CODEC according to claim 34, further comprising: an A/D clockgeneration unit generating timing signals for said A/D converter anddigital signal processor based on said transmission rate established bysaid analog modem.